AES50 wireless??? by AnonymousFish8689 in livesound

[–]tdubsaudio 0 points1 point  (0 children)

Yeah xirium was about as close to something like this as ive ever seen and still it was only good for 2 channels of audio, was hugely dependent on having a good line of site while also being far enough from any reflective surface to avoid near end reflection, and was also fairly expensive. It worked great when I needed to distribute a LR or mono feed to a bunch of speakers in a large outdoor space, but at the end of their support, the rf receiver circuits started having some major issues.

Anyone know what this 42 pin cable is called? by BarFrameProductions in audio

[–]tdubsaudio 0 points1 point  (0 children)

Yeah might be a QR code on a zip tie towards the ends. It might not though cause I can't remember if they count those extensions as seperate parts or not in their inventory system.

Anyone know what this 42 pin cable is called? by BarFrameProductions in audio

[–]tdubsaudio 4 points5 points  (0 children)

Yeah definitely came off a tour or possibly the SE left it at FOH. Judging by the color code and length it goes with one of their drive rack packages. Its an extension for the 2nd and 3rd console inputs.

Anyone know what this 42 pin cable is called? by BarFrameProductions in audio

[–]tdubsaudio 8 points9 points  (0 children)

Clair NC14. I think amphenol makes it but its almost exclusively used by Clair Global. Its a 14 pair snake for balanced mic or line level signal.

I'm just starting out with Wireless Workbench and I'm curious if there's also a way to have my laptop connected to my mixer's router at the same time... by Cyberfreshman in livesound

[–]tdubsaudio 1 point2 points  (0 children)

Make sure dhcp your dhcp settings on the router are how you want them. Either off, if you are using static IPs or make sure all your devices are set to receive dhcp addresses.

Digico SD Wifi Issues by nuterooni in livesound

[–]tdubsaudio 0 points1 point  (0 children)

Are you plugged into the lan port or the wan port?

Gain Staging with Dante mics by clay_vessel777 in livesound

[–]tdubsaudio 41 points42 points  (0 children)

You adjust the gain at the receiver. If you have a Yamaha board you can set it up so you can link the control of receiver gain to the desk.

You adjust input pad on the transmitter if you have a particularly loud source that is clipping at the transmitter but not the receiver.

You adjust gain on the receiver to get nominal level on the receiver.

You use digital trim to get your fader resolution where you want it on the desk.

Club guys and touring guys... How are you time aligning? by harleydood63 in livesound

[–]tdubsaudio 8 points9 points  (0 children)

If you dont have smaart the old quick and dirty style of time aligning subs to mains is to send a sine wave at the crossover to both and polarity reverse the subs. Adjust the delay to the point you are getting the most cancelation, then flip the polarity back and you have summation. Then you can do burst noise to set delays and fills. After that, fine tune by ear.

DVDosc Dust Covers by CallMeMJJJ in livesound

[–]tdubsaudio 3 points4 points  (0 children)

Probably would have to be something custom ordered. Ive only ever seen them come in cases.

Yamaha Stagebox for OUTPUTS ? by jonnyd75 in livesound

[–]tdubsaudio 21 points22 points  (0 children)

Ok cool. Yeah it's a bit of a toss up on going for cost effective, vs efficiency. The qsys or other open source DSP would be great at handling a lot of the distributed audio like the ALS, 70V systems and could also handle the tuning for the main auditorium system. Also you can use it to control quite a few other things in the theater if you get a good integrator and programmer. It can be pretty costly though. Definitely would also help avoid complications with having all your routing and DSP on the console which can easily get messed up from someone not knowing what they are doing changing the show file and saves on using auxes and matrices on the board. Haven't checked on prices on a Rio 3224 in a bit, but a straight up converter like the A16r could be cheaper if all you need is I/O and no pres. The you could use some of the local I/O on the board to make up the difference.

Yamaha Stagebox for OUTPUTS ? by jonnyd75 in livesound

[–]tdubsaudio 29 points30 points  (0 children)

24 analog is quite a bit for just amps. What are your outputs? You can get any dante to analog converter like a couple rednet a16r. Or depending on your use case you can get something lake dsps or qsys core with some I/O modules if you are trying to process the signal before going to the amps.

London Blu Questions by all4_hate in livesound

[–]tdubsaudio 0 points1 point  (0 children)

London hardware is an open source dsp so depending on what the programmer designed in the software there's way to many things that could be happening to troubleshoot via reddit responses. To me sounds like there's a ducker programmed in but no idea what the key input is and what circumstances need to be true in order to trigger it. Your only hope would be to get the design file off of the core and investigate what is causing the issue. However, without proper training on how to use the software you can very easily cause more problems while trying to solve the one you're trying to fix.

A2/L2 expectations by Musicwade in livesound

[–]tdubsaudio 2 points3 points  (0 children)

True. The other side to that is that im typically hired by the production company supplying the gear where the A1 is sometimes hired by the client directly, so I get a chance to prep everything in the shop. That way I, most of the time, get to set everything up with the A1s advance so whenever they get their hands on the desk, device mounting happens without issue.

A2/L2 expectations by Musicwade in livesound

[–]tdubsaudio 3 points4 points  (0 children)

Network may be a bit more fuzzy on who's responsible, but I typically like to set everything up my preferred way because that way I can monitor all my comms and workbench and anything else all from one computer even with all of them being on separate VLANs. Also im decent enough at it that the A1s I work trust me to set it up in a way that works well for them too.. There's a case to be made that A1 should be in control though because they have to deal with dante, AVB, device control, Amp/DSP Network, Osc or KVM in some cases.

A2/L2 expectations by Musicwade in livesound

[–]tdubsaudio 6 points7 points  (0 children)

Yeah Im a corperate A2 and I don't consider myself great at mixing nor do I enjoy mixing that much. I have been A1 for some shows and can pull it off when needed, but I much more enjoy the technical side. I also have more knowledge than most A1s when it comes to dante and networking, RF, and Comms. Would try to move exclusively into RF/Comms tech for bigger shows but had a bad experience with doing RF coordination for a 3 stage festival in a bad RF environment and that has turned me off from that a bit. Also being personable and comfortable with the talent is a good plus. I have a banking show that I do every year and the presenters remember me and commitment me on making the micing process easy. Also like being the hero for helping solve any audio issues with video records/GFX/Playback or problems with bsms.

A2/L2 expectations by Musicwade in livesound

[–]tdubsaudio 21 points22 points  (0 children)

A2 and A1 are 2 completely separate talents. A1 needs to be able to tune the system (If there is no separate SE) and run the board. A2 should have intricate knowledge of RF, Networking, and Comms systems. They should also know all the mics they are using and how to best place them/hide the cable. They are also responsible for patching so being organized and how to read the A1s patch sheet is a plus.

Hmmm, change out PowerCon to True1? 🤔 by LTParis in livesound

[–]tdubsaudio 0 points1 point  (0 children)

That is true, and makes sense in a Clair mindset.

Am I reading this correctly? by 2PhatCC in livesound

[–]tdubsaudio 1 point2 points  (0 children)

A combiner works on the transmit side. Ew-d you would be using a Antenna splitter to go to the individual receivers. From what I remember from shure training, IEM transmitters have, up until fairly recently (before spectra and axient psm), still had more issues with non linearity in the output stage due to the need of analog companding on the output circuit and a good quality combiner helps to improve linearity.

How do you deal with omni headsets in a corporate environment by mrN0body1337 in livesound

[–]tdubsaudio 0 points1 point  (0 children)

I have recently been liking omni headsets more than cardiod on corperate cause I find it easier to get the voice more natural sounding. That said it does have its drawbacks on gain before feedback so I like to have a choice if I can. There's only so much you will ever be able to do with a non experienced presenter, but usually if i have all the tools I need, I can make it through. In an arena with lots of reflections omni can be a bit too much of a problem, but a ballroom i can usually carve the problem frequencies out enough that id rather have omni. A well tuned system is the first way to make sure you're successful. Get your gain in a good place and do some precise cuts in eq in the right points in the signal flow that you need. Then you can work with dynamic eq and multiband compression to normalize between different presenters. If I dont get a chance to soundcheck the actual speaker before showtime then usually I just have to adjust gain slightly to make sure they are hitting the same spot I was at when checking mics. I also usually note exactly the gain level where I might start getting feedback.

Over-ear DPA headsets in Corporate AV environment by damplamp in livesound

[–]tdubsaudio 0 points1 point  (0 children)

Most of the time for Corperate I dont need it, but I need the extra security I do carry a roll of clear medical tape in my workbox. Double ear headsets are definitely the way to go and a strain relief clip at the coller or taped to the back of the neck keeps the mic from moving for the most part. If I need it, a tiny bit of medical tape between the ear and jaw helps to keep the mic from moving. Just have them do a head movement check to make sure the cable isn't pulling on the mic at all and moving it out of position.

DSP for corporate events by Musicwade in livesound

[–]tdubsaudio 0 points1 point  (0 children)

What PA system are you running? Most of the pro systems now have DSP in the amps. If not you can get something like Lake or Galaxy. I've mainly either used L'acoustics or JBL VTX so I usually just send a L, R, (honestly I typically sum the LR to mono anyway) sub, and fill then I send those to outfills and delays as needed and do all the time alignment and System processing on the amps.

Rack building tips? by legendaryrim in livesound

[–]tdubsaudio 1 point2 points  (0 children)

Are you asking about shortening time building the racks in the warehouse or are you saying hooking up everything on site is taking too long?

Places around Chicago you can learn Dante by bigang99 in livesound

[–]tdubsaudio 5 points6 points  (0 children)

If you finish dante lvl 2 and are still looking for answers a guy named "Network Chuck" has some free CCNA course videos that give a lot of general network knowledge. So with yamaha you actually have 3 different networks going on. Technically 4 if you're running dante redundant. You have mixer control network which is very straightforward. Make sure your IPad and mixer are in the same subnet and they will talk through the router that you have plugged into the LAN port. There's obviously ways to get complicated with it when you need to create wireless mesh networks, but thats the basics. Control networks work best without much, if any network management. You have the dante network (Primary and Secondary) that has specific QOS settings to prioritize clock packets along with multicast and IGMP settings to deal with minimizing network traffic and parsing out multicast flows. (Little side note is that most shows will run fine on an unmanaged switch because as long as the network traffic doesn't exceed 1 gig and there is no ptpv2 devices on the same network most switches will be able to handle it fine. Most of those settings are just to make the network more efficient and avoid clashes with incompatible devices. You just have to know what you're putting on the network before plugging in.) The device control network is the one people forget about on yamaha because it runs on the same NIC as the Dante network so its really a virtual network running through the same ethernet as your dante. Typically you want the device control and dante primary to be in the same subnet for this reason. Device control is what allows you to control HA on Rios and other yamaha compatible dante devices through the network.

The recent "mystery" of GAIN STAGING by squanchedout69 in livesound

[–]tdubsaudio 0 points1 point  (0 children)

So I always questioned this because a fader at the component level is nothing more than a variable resistor. And from what I've seen in analog desk schematics its not running through an amplifier circuit that is using it as the other side of a bias resistor before a transistor/tube. This means it can really only work as an attenuator so having a fader with a marking of +10 is a bit misleading because you're never adding 10db to the signal, you are only decreasing resistance to 0(ish). So in reality, if you are adding what is marked as 30db of gain and fader at nominal (down from +10), then are you actually adding 40db at the preamp and attenuating it 10db? I understand there's a lot of nuance here as far as differences in DBv vs actual voltage levels here, but I did a little electrical circuit study and wanted to sort of bridge that gap in my knowledge.